Hi all, I can't find any material about impulse reponse normalization for a convolution reverb. Using Logic's space designer I notice that there's definitely a preprocessing of the impulse reponse that one loads: given the same input and impulse without preprocessing, the convolution would yield a maximum floating point value of 4 that will cause digital clipping. I can imagine that to avoid clipping for an arbitrary input, the normalization has to be done on the peak of the absolute value of the frequency response, right? But I have troubles figuring out how to look for this peak that theoretically can be anywhere during the decay of the impulse response: imagining a sliding window FFT analysis still puzzles because when I look at the output of the matlab command FREQZ(B,A,N) with varying N, I get naturally different peaks due to the different interpolations. Moreover, if this is the way to go, I wonder if the maximum part size of the partitioned convolution algorithm shall be used to set the size of the sliding window during the analysis of the impulse response. Thanks! Alessandro -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
