On Sun, Sep 11, 2016 at 04:14:33PM -0400, Ivica Ico Bukvic wrote: > Does anyone have access to an easy to implement (as in code) > dereverberation algorithm assuming one has access to the room > impulse response? I am looking for a way to clean-up recorded audio > signal in order to improve signal clarity (e.g. speech). Any info on > this topic is most appreciated. This is not an easy matter. There is no single 'room impulse response', there is a different one for every combination of source and receiver position. And even these can't be assumed to be constant except maybe at low and low mid frequencies. So in practive any given room IR is just a hint, and algorithms that work need to be adaptive. They usually work in the frequency domain, and build up statistics of dynamics in each frequency band. Using these it's possible do attenuate some of the reverb. How much depends strongly on the source material. This is still a very active research topic. Most research is related to speech pickup, not music. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) _______________________________________________ Linux-audio-user mailing list Linux-audio-user@xxxxxxxxxxxxxxxxxxxx http://lists.linuxaudio.org/listinfo/linux-audio-user
