scott@heim.glendale.ca.us (03/16/90)
In article <8608@chaph.usc.edu> alves@nunki.usc.edu (William Alves) writes: >In article <2441@rodan.acs.syr.edu> jmwobus@rodan.acs.syr.edu (John Wobus) writ\ >es: >>I am having fun writing a C program that manipulates files of sampled >>audio, but have been mulling over the best way to take data sampled >>at a high rate (say 55000 samples per second) and producing data >>sampled at a lower rate (say 11000 samples per second) with a minimum >>of distortion. Can anyone point me to any simple and/or efficient >>algorithms or strategies to do this? >> >The intuitive response would be to take every fifth sample. However, doing >that without some kind of filtering leaves you open for all kinds of aliasing >problems. Thats right - that's why the the device which did the original sampling probably had a lowpass "anti-aliasing" filter infront of the A/D. You can simulate this same procedure in software (for an 11K sample rate, you probably want to pass 5K and below), however I don't have a lowpass code fragment for you. Maybe some kind comp.music reader can help out - otherwise you might want to address this to comp.dsp. -- Scott Watson - "Inane little message goes here" uucp: {rutgers,ames}!elroy!grian!heim!scott Internet: scott@heim.glendale.ca.us