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Audio Weaver contains several filter types which operate on raw coefficients. These filters are for expert users who understand DSP and know how to calculate the filter coefficients[1]. There are two types of filters – Finite Impulse Response (FIR) and Infinite Impulse Response (IIR). Although Audio Weaver supports both types of filters, the majority of the filters used in audio applications are IIR due to their computational efficiency.
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There are 5 coefficients that the user must set: b0, b1, b2, a1, and a2 (a0 is always assumed to be 1). Audio Weaver does not check for stability and care must be used when computing the filter coefficients. There are several variants of Biquad filters. The simples – Biquad – has a single stage and implements the different equation shown above. BiquadCascade implements N stages of filtering with each channel using the same coefficients. BiquadNCascade implements N stages with each channel have its own set of coefficients. Finally, BiquadSmoothed implements a single Biquad stage with coefficient smoothing on a block-by-block basis.
  | FIR
| Time domain FIR filter Specify filter length in module properties | |||||||||||
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 |  | Biquad
| Second order IIR filter. 5 filter coefficients are specified. No smoothing. | ||||||||||
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  | BiquadCascade
| Multiple Biquad filters in series. The number of filters is specified in module properties. The same coefficients are used per channel. | |||||||||||
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|  | BiquadSmoothed
| Second order IIR filter. 5 filter coefficients are specified. Smoothed on a block-by-block basis | |||||||||
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| BiquadNCascade
| Multiple Biquad filters in series. The number of filters is specified in module properties. Different coefficients are used per channel. | |||||||||
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| FIR Sparse
| Sparse FIR filter in which most values are zero. Less convolution cycles than normal FIR |
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 | FIR Sparse Reader
| Sparse FIR that connects to a delay state writer. Convolution is based on a pointer rather than a separate FIR buffer.
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FIR Sparse Reader Fract16
| Like FIR Sparse Reader except half the memory. Data is converted to fract16 for computations and has a conversion for the output if necessary. |
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This module contains several SecondOrderFilterSmoothed modules in series. This can be used to implement a more complicated EQ with only a single module. Under module properties, specify the number of stages of filtering. If the number of stages is set to 1, then this module is equivalent to the SecondOrderFilterSmoothed module. When there are multiple stages, the inspector expands as shown right:
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[1] Matlab is often used by expert Audio Weaver users to compute coefficients and then update them in the block diagram.