Description:

The filters created here will be directly applied to music signal after it is loaded into convolver software.

Options:

1. If there are digital crossover filters in the workspace, this is automatically recognised when the filters are created.
2. The directory where the impulse response recorded with the LogSweep Recorder is located.
3. The window for Excess Phase correction (Order: Left Low/High, Right Low/High¹ ).
4. Adjusts the level in Macro 5 between the recorded impulse response and the corrected response to facilitate visual comparisons.
5. The level of the filter can be adjusted with this value (Caution: Exceeding 0dB may produce clipping).
6. Correction filters are created for the selected sample rates
7. Enables the correct creation of filters with a sample rate higher than the measured impulse response.
8. Creates a filter as a 32-bit .WAV or 64-bit .WAV (deprecated in newer versions of Acourate).
9. Pre-ringing compensation. Several iterations between Macro 4 and Macro 5 are required to check for presence of pre-ringing.
10. If a subsonic filter is desired, enter the cut-off frequency here.

¹ : The low value is 20Hz, the high value is 20kHz. The time values in between are adjusted evenly by Acourate.

Use:

After defining the parameters such as filter sample rate and subsonic filter, the correct values ​​for excess phase correction must be determined experimentally. Since these vary depending on the loudspeaker/room and setup, several iterations are necessary between Macro 4 and Macro 5 (test convolution). In each iteration, enter new values ​​in 3 Excessphase Window / 9 Pre-ringing compensation and examine the result with a test convolution.

It therefore makes sense to first only tick the box for 6 Filter sample rate for the sample rate used in the measurement (48). Once the optimal values ​​have been found after a few test convolutions, the filters can be created for all required sample rates as the final step.