StandardCLIP Manual

Introduction

StandardCLIP is a new advanced clipping-plugin. You can use StandardCLIP in two ways:
  • as a dynamic tool in your master effects chain to increase the volume
  • as a harmonic generator to add odd harmonics to your signal
You can adjust the way the clipping is done easily, like a hard-limiting brick wall or smooth soft-saturated.
Since clipping is a non-linear process, clipping adds always new harmonics to the signal, and if it's done without oversampling, harmonics above the Nyquist-frequency will be folded back in the frequency spectrum and that's called aliasing. Oversampling reduces aliasing. StandardCLIP offers high quality oversampling up to the factor 256. It also offers very versatile oversampling options: You can choose between minimum phase mode (without additional latency) and linear phase mode (adds latency to the signal, but preserves phase-shift), and change the filter quality and cut-off frequency of the filter which is used for the oversampling process.

Why Oversampling?

This is a 2000Hz sinus signal running through the clipping unit without oversampling.
Clipping without Oversampling
What you see is that odd harmonics (3rd = 6 kHz, 5th = 10 kHz, 7th = 14 kHz) are added to the signal, this is what naturally happens when you limit any kind of signal.
But we also see unwanted side effects, called aliasing. Because the new odd harmonics will also continue beyond the Nyquist-frequency. These harmonics can not be recreated with the used sampling rate and will be folded back to the signal. StandardCLIP reduces the aliasing by oversampling the source signal to a higher sample rate. Then it processes the clipping, removes unwanted frequency content above the source sample rate and finally down-samples it to the source sample rate again.
Clipping with Oversampling
This is another example of clipping with and without oversampling. The source signal is a sine sweep from 800Hz to 20kHz.



The quality of the oversampling depends not only on the oversampling-factor. Also the quality of the filters which are used for up and down sampling is essential (steepness, phase linearity).
StandardCLIP offers total control over the used filters.

Signal-Diagram

Signal Diagram

Installation

Mac

VST & AudioUnit
Open the DMG-Package and copy the included
StandardCLIP.component into the folder /Library/Audio/Plug-Ins/Components/
StandardCLIP.vst into the folder /Library/Audio/Plug-Ins/VST

If the plug-in should be only accessible by one user-account, then copy the files into the folders
/Users/username/Library/Audio/Plug-Ins/Components
/Users/username/Library/Audio/Plug-Ins/VST
instead.
Pro Tools AAX
Move the AAX-file into "(Macintosh HD)/Library/Application Support/Avid/Audio/Plug-Ins".

Windows

VST
Copy the DLL-File into your VST-plugin folder. If you are not sure where the VST-plugin directory of your VST-host software is located, please look into the manual of your VST-host software.
The plug-in comes in two different versions for windows, a 32-bit (x86) and a 64-bit (x64) version. Please be sure that the host is using the same architecture as the plugin.
Pro Tools AAX
Move the AAX-file into Pro Tools Plugin-Folder. This is usually C:\Program Files\Common Files\Avid\Audio\Plug-Ins.

Enter License Key

First time you open the full version of StandardCLIP, a license-key dialog will appear. Please enter the license key you received and restart your host application.

GUI Overview

Clipping without Oversampling

Parameters

Main

Input-Gain
Amplification before the clipping stage.
Increase the input gain to drive the signal into the clipping barrier.
Clip-Level
Adjust the clipping level.
Output-Gain
Amplification after clipping stage.
This is the main output-volume control. If you use the StandardCLIP as last plugin in your signal-chain, the computed clip-level (clip-fader * output gain fader, small red arrow) should be always 0 dBFS or lower. Because of the oversampling process, signal peaks higher than computed clip-level can occur.
Output Gain Assistant
The output gain assistant helps you find a reasonable headroom between the computed clip level (clip-fader * output gain fader, small red arrow) and the ceiling level.
Due to the oversampling process, some signal peaks can appear above the computed clip level. In order to avoid aliasing, it is important that the signal level never overshoots the ceiling level.
This assistant adjusts the output gain fader to the highest possible value, which guarantees that no output sample level will be higher than the ceiling level. Please play the loudest part of your project while the assistant is open.
Click on "Apply New Output Gain" to set the new output gain.
Ceiling
Use ceiling only if you want to use StandardCLIP as final plugin of your master chain, to prevent samples higher than a defined threshold which is usually under or equal 0 dBFS. The ceiling threshold should never be lower than the computed clip level (small red "CLIP" arrow), because the ceiling occurs after the oversampling stage and will create aliasing then. Usually the ceiling is just a few dBs higher than the computed clip level, to limit the few samples above output-gain which can occur because of the oversampling process.
Clipping mode
StandardCLIP has a switch to select several different clipping modes.
Soft Clip Classic
This is the default algorithm, and it’s the same one, which was used in the previous version.
Soft Clip Pro
This new algorithm has a slightly different characteristic. It compresses an adjustable dynamic range by half. A line in the function plot shows from which level the saturation begins. The dynamic integrity below this threshold stays intact.

25% - compresses the last 6 dB into 3 dB, and adds 3 dB
50% - compresses the last 12 dB into 6 dB, and adds 6 dB
75% - compresses the last 18 dB into 9 dB, and adds 9 dB
100% - compresses the last 24 dB into 12 dB, and adds 12 dB
Hard Clip
cuts the audio at the clipping level without soft-saturation
Soft Clip Saturator
Intensity of soft-clipping function. You can adjust seamlessly between hard-clipping and soft saturated clipping.

Oversampling

Use oversampling to minimize aliasing artifacts.
The higher the oversampling factor, the more you reduce aliasing caused by the clipping process. But it will also cause more CPU consumption.

To get an impression how the oversampling affects the processing, here are some frequency spectrum-graphs. A two kHz sinus-test signal, which is processed by StandardCLIP with different oversampling factor settings.
Oversampling Aliasing
Oversampling-Advanced settings
To open the advanced settings, click on the oversampling-button -> Expert settings, or use the Preferences button on the top -> Oversampling Configuration
Oversampling Options
Band-limiting Filter type
The filter is an important part of the up and down-sampling process. You can choose between to filter-types:
  • linear-phase (adds latency to the signal, but doesn’t induce phase shifting to the signal. This type is preferred for most applications (inclusive mastering)
  • use minimum-phase filter (adds no latency to the signal, but induces phase-shifting to the signal) for live usage
Oversampling Factor Offline
This can be used to have a higher oversampling factor while bouncing (whether or not this is supported, depends on the host). When using an offline oversampling factor, which differs from the online oversampling factor, the online oversampling factor has to be at least 2. This is important for various reasons. (Among other things, online/offline have to share the same latency setting)
This function has been successfully tested with:
  • Pro Tools 12 (File -> Bounce To -> Disk), the „AudioSuite“ clip rendering function from the menu still uses the online oversampling factor
  • Logic Pro X (File -> Bounce -> Project or Section)
  • Cubase 8 (File -> Export -> Audio Mixdown)
Filter Cut Off
The lower the filter-cut off the less harmonics above Nyquist-frequency will cause aliasing, but the more effect on the signal the filter has.
Filter Kernel Size (only with linear phase filter)
Generally speaking the steeper the filter is, the better the oversampling result will be. But keep in mind that bigger filter-kernels also cause more latency, CPU consumption and potential pre-ringing artifacts in the transitional region. On the other hand, bigger filters have also a smaller transition region.
Transition Region
Filter-Order (only with minimum phase filter)
The higher the order, the steeper the oversampling filter, but also the higher the CPU-consumption. Also there will be more phase-shifting in the transitional region.
Reset
This button resets all oversampling related settings to their default value.

Additional Parameters

Active Channels
Active Channels
Some host-software will not open the plugin with the right number of channels, which will waste CPU-power. With this option you can adjust the right number of channels the plugin should use.
Silence Bypass
To save CPU-consumption the silence bypass will be automatically activated if the input audio level is below -190 dB. You can turn off this behavior via the options menu.
Silence Bypass
Zoom
Adjust the whole plugin-GUI size.
Zoom

Displays

Function Plot
StandardCLIP visualizes the clipping function as logarithmic and linear function plot. To switch between the linear and logarithmic function plot, click on it. A small red point in the diagram shows the current peak-value.
Function Plot Logarithmic
Function Plot Linear
Waveform Display
This is a preview of how the output or input waveform looks like, using the maximum level of all channels as input source.

The output preview includes the pre-reduction level (red), relative to the output-level (white). If you use no soft-clipping the maximum output-level (white) will match the clipping level (If output gain is 0dB). The more soft-clipping is used, the higher the pre-reduction level (red), because of the inclusive amplification of the saturation function.

The thin line represents the 0dB boundary.

You are able to switch between a full waveform preview or a half waveform preview, for a more precise overview.

Its also possible to switch between slow and fast scrolling.

Waveform Display
Max Peak. and Max. RMS (Root Mean Square)
Shows the all-time maximum peak and maximum RMS level. You can reset both levels with a mouse click.
Input/Output Meters
Meters
This is a peak-meter, a peak-hold meter and an RMS-meter. Simply control the input/output gain of your signal. A big red block on the top signals the output going above 0 dBFS. This is especially important if you use StandardCLIP as the final plugin in your mastering chain.

Usually RMS should be always lower than -12 dBFS (for modern pop-production) to still produce a punchy dynamic mix. Except if you are David Guetta or Metallica. Change the range of the scale via the options menu.
Gain Difference Meter
The Gain Difference Meter shows the RMS (red, Root Mean Square) or peak (green) difference between the input and output of the oversampled clipping process. The output gain and ceiling level have no effect on this display. Change the minimum and maximum scale values via the options menu.
Gain Difference Meter

Credits

Special thanks to all beta-testers and to Matthias Kahlmann and Conor Boyle for having helped me with the manual.
Copyright 2016 Christian Knufinke. VST is a trademark Copyright of Steinberg Media Technologies GmbH. All other trademarks and copyrights are the property of their respective owners.