Hello. I'm particularly interested in improving audio compression using open standards (being a classical record producer, sometime radio and tv producer and practicing musician) though I believe that in a few years, we'll have better data storage and transfer facilities that will render much compression un-necessary. Still, for the time being, let's make it sound as good as possible! Using the latest snapshot of pgcc for Linux (071598), I found that compiling with -ffast-math caused an illegal MPEG3 bitstream to be output, and compiling with anything more than -O2 (and -march=pentium, -mcpu=pentium) caused the program to segfault (I'll gdb it later to tell you exactly where). I found that encoder worked in mono (altering the symbol in main.c from MODE_STEREO to MODE_MONO) encoding a mono .wav file, and made a good job of the sound down to 40kbs. By the way, the sound was much better encoding a 32kHz sampled .wav at these rates than encoding a 48kHz .wav (same material, converted by the 'sox' program) which is my audio Swiss army knife. If I can help implementing/testing differeny psycho-acoustic models, and particularly the MPEG-II layer III stuff, I'd be only too happy to help. My experience in programming is limited (though I can understand the code OK), debugging slightly better, using the GNU compilers under Linux is fine, and I can do lots of listening! This is the studio where I work. John Hayward-Warburton [EMAIL PROTECTED]
