If I compile an IR-filter with fs set to 192 kHz the underlying biquad parameters will be optimized such that I get the required filter response for any signal that has a sampling frequency of 192 kHz.
If the same filter is used with a signal with 48 kHz sampling frequency all filterfrequencies will be effectively shifted downwards by a factor 4.
To get the proper filter response the schematics have to be compiled with an fs of 48 kHz. This will give me the proper biquad parameters for this specific sampling frequency.
It’s all easy and logical.
My problem: When I use the Reverb algorithm in combination with the ADAU1463, then the parameters of the algorithm are independent of fs!! Whether the fs is set to 48 or 192 kHz makes no difference.
As a result, the reverb signal calculated for an “input” signal with a sampling frequency of 192 kHz will be different from a signal with the same shape but with a sampling frequency of 48 kHz. With the latter the reverb signal is more delayed in time (4x) and is more spread in time (four times wider).
So my questions:
+ For what sampling frequency has the reverb algorithm been optimized?
+ Can the parameters of the parameters access table of the reverb algorithm (coeffs, dif_cos, dif_sin, ….) been optimized for different sampling frequencies?
+ Is there any documentation available what all these parameters do?
Currently I’m working on an audio processor that does all the processing at the default frequency of the incoming signal. The reason is, that I’ve found the asynchronous sample rate converters to have a slight muddying effect on sound quality.
I want to add reverb to the processor but this is only possible if the response can be made consistent for all sampling frequencies.
Thanks in advance,