Normally the phase will be varied with the change of gain when using filter in sigmastudio, but if phase variation too much will impact audio performance. so is there any filter in sigmastudio avaiable to keep phase unchanged when gain is changed?

Normally the phase will be varied with the change of gain when using filter in sigmastudio, but if phase variation too much will impact audio performance. so is there any filter in sigmastudio avaiable to keep phase unchanged when gain is changed?

Hi Ben,

The standard audio filters in the SigmaStudio library are 2nd order biquad IIR filters. They have a non-linear phase, as you mentioned.

If you want to implement a linear phase filter, then the best option would be to use the FIR filter (also available in SigmaStudio). However, SigmaStudio does not include FIR filter design tools, so you would have to use external software (Matlab, for example) to generate the coefficients.

Hi Brett and Ben,

*Dispro*is a legacy filter design tool to figure IIR and FIR coefficients. Its author has graciously made it available for free at http://www.digitalfilterdesign.com/ It runs in a DOS window under XP and Vista, or via a DOS emulator (such as the open-source DOSBox) under Windows 7. The screenshot below shows the FIR coefficients it calculates for a bandpass filter centered at 2KHz:A FIR filter's list of coefficients (its

*kernel*) directly describes the filter's impulse response. As shown above, coefficients #1 and #59 are the same, as are #2 and #58, etc. This creates a left-right symmetrical impulse response, with its resulting linear-phase characteristic. Unfortunately, this 2KHz filter needs 58 delays and 59 multiplies -- and it gets worse at lower frequency. By comparison, IIR bi-quad filters need only five coefficients and two delays, thus find more use in common audio applications.Bob

Hi Brett and Ben,

Disprois a legacy filter design tool to figure IIR and FIR coefficients. Its author has graciously made it available for free at http://www.digitalfilterdesign.com/ It runs in a DOS window under XP and Vista, or via a DOS emulator (such as the open-source DOSBox) under Windows 7. The screenshot below shows the FIR coefficients it calculates for a bandpass filter centered at 2KHz:A FIR filter's list of coefficients (its

kernel) directly describes the filter's impulse response. As shown above, coefficients #1 and #59 are the same, as are #2 and #58, etc. This creates a left-right symmetrical impulse response, with its resulting linear-phase characteristic. Unfortunately, this 2KHz filter needs 58 delays and 59 multiplies -- and it gets worse at lower frequency. By comparison, IIR bi-quad filters need only five coefficients and two delays, thus find more use in common audio applications.Bob