FabFilter

Hi Mark,

We prefer to keep the exact algorithm that we use for linear-phase processing to ourselves :-). We'd like to encourage people to listen to our plug-ins and judge them by the actual sound quality and "feel" when working with them, instead of by theoretical aspects of the algorithms involved.

The filter algorithms in Pro-Q are special in the following ways:

1) They have a very smooth and quick response when changing the frequency and other settings of an EQ band, while most other EQ plug-ins either let you choose from a set of fixed frequencies or have to "reset" themselves each time you change a parameter. This gives a much nicer behavior, making it much easier to narrow down on a specific frequency that you'd like to suppress, for example.

2) The filter frequency is not limited by the sample rate divided by two: you can actually place a bell filter at 30 kHz when running at a sample rate of 44.1 kHz and still have it influence the sound exactly like an analog filter would. Also, we achieve this without internal oversampling, so it doesn't introduce any latency in zero-latency mode.

3) The ability to run in exactly the same way in zero-latency and various linear-phase modes.

We think it's the combination of these features that make the filters stand out from the crowd...

Cheers,

Thanks for the response.

So Pro-Q does not have filter pre-warping problems around the Nyquist when working at lower sample rates?

You did demo the Pro-Q right Mark?

Hi Mark,

Indeed there are no pre-warping problems. Also, I forgot to mention that Pro-Q realizes all of this without excessive CPU usage. :-)

Cheers,