Changing filter's coefficients to compensate sampling rate changes 2009-11-22 22:24:31 Hello,I've read the Recommendation ITU-R BS.1770.1, titled "Algorithms to measure audio programme loudness and true-peak audio level", and there is one sentence which makes me doubt. It refers to the coefficients of one filter (2nd order IIR) used in the process of measuring the loudness level of an audio signal, and says: "These filter coefficients are for a sampling rate of 48kHz. Implementations at other sampling rates will require different coefficient values, wich should be chosen to provide the same frequency response that the specified filter provides at 48 kHz"I've been looking for methods of adjusting coefficients from a given change in the sampling rate, but all I've found involves polyphase filters (which means adding more filters instead of modifiying the coefficients of an existing one). Besides, if I've understood this stuff, in the Z transform, a change in the sampling rate is the same as a multiplying the exponent of the exponential by some factor, so, how a change in the filter coefficients (constants multiplying the exponentials in the Z transforms) can make the same effect as some changes in the exponents of the exponentials? Obviously there is something I have mistaken or something I don't know. In either case, which really matters to me is how can such change in the filter's coefficients be made (if there is any method or something similar) to get the same (or similar) frequency response?Thank you very much.