RE: [linux-audio-user] Announcing Audio* Music Synthesis Project

Okie dokie, there is no rush, I'm not going anywhere hehe. I'm still working away on it each day. I didn't go to school to make synthesizers so I learn the only way I know how, by experimenting and reading everythign I can find about it. You can hear a sample I did, ignore the initial click, it doesn't that in real-time. I'm figuring out why it does that. But Port Audio is really not hard to compile and get working either. Just run the make -f Makefile.linux, and copy the libportaudio.so to your /usr/lib directory. FLTK is a little harder though, but maybe CCRMA planet thing has it. I link statically because I use some cool widgets I found from another project that won't compile yet. Well good luck and all and thanks for interest it is lonely no one has shown any interest yet but I've only been working on it for a month or two. I have made alot of effort to make sure it does not alias and the filters sound good. Really, the filters are the most important part, so I did alot of hard work on them. A real analog synth will have variations in the waveform, but I have experimented, and it doesn't make that much of a difference to be honest. I tried simulating condensor unloading by different means to round off the waveform, it really didn't make any difference. I made variations on the amplitude and clipping, not much of a difference. I tried clipping the sawtooth to get a 303 style saw, didn't make that much difference, but it sounded more squarish. I think the filter is really what makes an analog synth sound so good or bad. I want to translate a Curtis VCF into LaPlace formula and than bilinear transform it into Z domain (digital). I found a cool paper on the moog, where he did the hard work to make the Laplace transform already, and I just did that with bilinear into digital. Than I double sampled it to get it stable. It sounds pretty close to a real Moog filter but not exactly ( I had the real thing). The hard part was figuring out I had to double sample it (resonance would blow out the filter otherwise). Also, it didn't work with 32-bit floats correctly, but I notice no difference in using doubles anyways, other than twice the memory requirments which reminds me I need to look and see how much memory it is using. Eeek! I forgot to only make 1 static buffer for the waveforms, it was taking 230 mb of RAM!! I just fixed it it should only take about 32mb. Ok now it's fixed, I forgot about that... See yah!