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| WaveWarp 2.0 Component
      
Digital Filters:
Functional Description
Finite Impulse Response (FIR) digital filter
employing fast (FFT-based) convolution for the filtering computation.
This is specially geared for large filter lengths (e.g. for room simulations etc).
The filter can be designed off-line using a third-party application
(such as
MATLAB®
or
QEDesign®
), then imported to WaveWarp as an ASCII file via a simple dialog box
in the Parameter Window
(refer to the
WaveWarp Users' Guide
for
a detailed description of the specific FIR structure implemented in WaveWarp, and the
associated ASCII file format for storing the coefficients). For MATLAB users in
particular, the "wwmatlab" sub-directory of the WaveWarp root directory contains
the necessary function m-files (plus example scripts) for exporting
filters from MATLAB, enabling the seamless integration of
MATLAB's powerful filter design tools with WaveWarp's real-time audio engine.
(refer to the
WaveWarp Users' Guide
for
a summary of all bundled m-files for working with MATLAB
in a variety of areas in addition to digital filter design).
The "Gain" slider(s) adjust(s) the signal output level(s) after filtering.
If the application specifically requires selective filtering at low frequencies, it is recommended to
employ multirate techniques, whereby the signal is downsampled before filtering.
The Multirate
category of the Component Library contains a wide range of downsamplers (and upsamplers) for this purpose.
All WaveWarp components automatically adapt to the sample rate of the incoming signal, so it is straightforward
to connect a digital filter component after a downsampler in order to realise the significant performance
gains inherent to multirate techniques
(refer to the
WaveWarp Users' Guide
for more information on WaveWarp's multirate signal processing functionality; and see
[CrRa]
and
[StNg]
for a detailed treatment
of multirate filtering.)
Algorithm
The FIR filter is implemented using straightforward fast convolution, so the
latency will always be equal to the filter length, regardless of CPU speed.
If latency is unacceptable, then use the "Large FIR (Hybrid)" component which utilises
a proprietary algorithm to achieve the same result but with zero latency
(though at the price of higher computational expense).
Signal Implementations
| Audio signals | Control signals | Description |
|---|
| Single input single output mono-mono | n/a | The mono audio input is filtered and sent to the mono audio output. |
| Single input single output mono-stereo | n/a | The mono audio input is filtered and sent (in duplicate) to the stereo audio output channels. |
| Single input single output stereo-mono | n/a | Each audio input channel is filtered separately (but with the same filter coefficients). The filtered channels are then averaged and sent to the mono audio output. |
| Single input single output stereo-stereo | n/a | Each audio input channel is filtered separately (but with the same filter coefficients) and sent to the separate stereo output channels. |
Related components:
Example DrawingBoards illustrating usage:
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