Yet another plugin announcement, I’m happy to announce the release of a chorus plugin as well as a stereo phaser plugin based on the Audio Toolkit. They are available on Windows and OS X (min. 10.8) in different formats. The chorus plugin is based on a variable delay filter driven by a random delay signal, and the stereo phaser morphs a mono signal in a stereo phased pair.

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I’m happy to announce the release of a variable delay plugin and an update to a fixed delay plugin based on the Audio Toolkit. They are available on Windows and OS X (min. 10.8) in different formats. The variable delay plugin allows to define a delay based on a sinusoid (mean delay, delay amplitude and frequency) and mix it with feedback and forward signals. The fixed delay plugin allows a fixed delay (up to a second). Both plugins can run at 192kHz.

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I’m happy to announce the release of a side-chain stereo expander based on the Audio Toolkit (yes, after a compressor it is time for an expander!). It is available on Windows and OS X (min. 10.8) in different formats. This stereo expander can work on two channels, left/right or middle/side, possibly in linked mode (only one set of parameters), and can be set up to mix the input signal with the compressed signal (serial/parallel compression). The side chain channels can be used to steer the gain stage (the same setup will be used, right/left or middle/side).

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There are several different low pass filters, and as many high pass, band pass, band stop… filters. In Audio toolkit, there are different usual implementation available:

  • Bessel
  • Butterworth
  • Chebyshev type 1
  • Chebyshev type 2
  • Second order
  • Linkwitz-Riley
  • RBJ

and it is possible to implement other, different orders as well…

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I’m happy to announce the release of a side-chain stereo compressor based on the Audio Toolkit. It is available on Windows and OS X (min. 10.8) in different formats. This stereo compressor can work on two channels, left/right or middle/side, possibly in linked mode (only one set of parameters), and can be set up to mix the input signal with the compressed signal (serial/parallel compression). The side chain channels can be used to steer the gain stage (the same setup will be used, right/left or middle/side).

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Convolution is an algorithm that is often used for reverberations. If the equation is easy to understand and to implement, the implementation is costly. The other way of doing it is to use Fast Fourier Transform (FFT), but the direct/crude implementation requires latency. If it is possible to optimize the basic convolution code, it is sometimes more interesting to use a different algorithm, as it is the case here.

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When I first read about transient shaper, I was like “what’s the difference with a compressor? Is there one?”. And I tried to see how to get these transient without relying on the transient energy, with a relative power (ratio between the instant power and the mean power) filter, or its derivative, but nothing worked. Until someone explained that the gain was driven by the difference between a fast attack filtered power and a slower one. So here it goes.

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