am42+@andrew.cmu.edu (Alexander Paul Morris) (10/26/90)
I'm interested in taking a digital sound file saved on a PC and adding (digitally, by means of a program) effects such as echo, reverb, delay, chorus, distortion, etc. Does anyone possibly have any code or algorithms they could share with my that may facilitate my ability to do this? Also, would anyone know about any books that may be published that might describe how a computer program could add such effects to a digital sound file? I have two books: Digital Audio Signal Processing and An Intro to Digital Signal Processing, but they really don't give any useful information on implementing any effects such as those described above. Any help will be VERY MUCH appreciated. I've tried to find information on this for a long time. Alexander Morris Carnegie Mellon
lsalomo@hubcap.clemson.edu (lsalomo) (10/26/90)
I have a book titled "Musical Applications of Microprocessors", which (among the MANY other GREAT topical discussions) describes such as what you want to do written in 68000 assembler and BASIC (the only detracting thing of the entire book). I'll see if I can remember to find out what the ISBN number is and post it. Cheers, Q - the "Q"uestor for knowledge (, a degree, etc.) lsalomo@hubcap.clemson.edu ibmman@prism.clemson.edu ibmman@clemson.clemson.edu ============================================================================= "Gee Wally, I think there's something wrong with the Beaver." =============================================================================
bkuo@nunki.usc.edu (Benjamin Kuo) (10/27/90)
Echo, reverb, and delay are all subsets of just one function -- taking the original sound and repeating it at correct intervals... They are pretty easy to implement, it simply involves taking your waveform, feeding it to a function, mixing in the sound, and progress with lower and lower levels back into the function. There is a book "Musical Applications of Microprocessors" published in 1985, which goes over the basic ways to modify sound. It's a bit outdated (!) now with all the DSP and other neat digital tools available, but it covers the basics quite well... I think it is published by Hayden Books. (Now if you had a Mac, I would just direct you to any sound editing program, where that stuff is pretty standard... Effects, filters, other neat tools...) Benjamin Kuo
broehl@watserv1.waterloo.edu (Bernie Roehl) (11/01/90)
In article <gb9uE9_00Uh_E4J58s@andrew.cmu.edu> am42+@andrew.cmu.edu (Alexander Paul Morris) writes: > >I'm interested in taking a digital sound file saved on a PC and adding >(digitally, by means of a program) effects such as echo, reverb, delay, >chorus, distortion, etc. Does anyone possibly have any code or algorithms >they could share with my that may facilitate my ability to do this? It's actually pretty easy to do most of those things. You essentially add the waveform to itself, sample by sample, with an offset and a weight. Your first output value is the first value of the input. Your second output value is your second input value plus half the first input value. Your third output value is half the second plus a quarter of the first. And so on. out[0] = in[0]; out[1] = in[1] + .5 * in[0]; For all subsequent indices (2, 3, 4 etc): out[i] = in[i] + .5 * in[i-1] + .25 * in[i - 2]; (The above assumes you're storing values in floating point, which is a bad idea; the principle is the same, regardless. And needless to say, you can use more that just the previous two samples). Hope this helps. -- Bernie Roehl, University of Waterloo Electrical Engineering Dept Mail: broehl@watserv1.waterloo.edu OR broehl@watserv1.UWaterloo.ca BangPath: {allegra,decvax,utzoo,clyde}!watmath!watserv1!broehl Voice: (519) 885-1211 x 2607 [work]
jensenq@iconsys.icon.com (Quinn Jensen) (11/02/90)
Look into csound from the MIT media lab on ems.media.mit.edu. It does do reverb and other effects on sound files. The source code is supplied. Perhaps more importantly to you, they do document their signal processing algorithms.
Peter.Becker@f6.n492.z5.fidonet.org (Peter Becker) (11/18/90)
Can't really help - but have you ever seeen a program by the name of mushroom that went around some time ago? It used a huge file of digital samplings to produce quite good sound (for a PC speaker) for about 20 secs. You may be able to get some techniques from that. -- Peter Becker - via FidoNet 5:494/5 UUCP: uunet!m2xenix!quagga!bberry!492!6!Peter.Becker ARPA: Peter.Becker@f6.n492.z5.fidonet.org
Peter.Becker@f6.n492.z5.fidonet.org (Peter Becker) (11/18/90)
DIstortion would be achieved by applying a maximum cutoff to the sampled waveform. To acheive a richer effect, you could use Fourier analysis to decompose the waveform into componenet frequencies, and apply different cut off levels to each frequency range. -- Peter Becker - via FidoNet 5:494/5 UUCP: uunet!m2xenix!quagga!bberry!492!6!Peter.Becker ARPA: Peter.Becker@f6.n492.z5.fidonet.org