hi there, hope this mail is on topic on this list. for an FM and internet radio project during theworldcup 2006 in Berlin, we're looking for support onthe linux audio developer side. it's a cultural projectand we'll recording audio events at differentplaces, from clubs to universitities. there's a small budgetincluded and some preliminary code is available froman older streaming project.of course the project will be released under GNU.if you are interested to join in and help us with yourwork, please contact us quickly at:kontakt->radioeinszueins.de here's the basic setup we're working on,it derives from years of practical d.i.y.experiences with recordings and streamingof club events. your questions and comments are welcome. ..................................................................................................................... >> automatic event recorder: questions/tasks:- how to profit from 24bit to 16bit dithering and 88.2 khz to 44.1 kz resamplingfor dynamic limiting/compression aka "Automatic Gain Control"?- silence detection (pause of recording or auto cutting)- premastering to have recordings broadcast-ready and normalized(up to the much hated brickwalled optimod FM sound..)- tradeoff between sound quality and throughput- presets for speech, music (jazz, classic, rock, electro), room micros+ making podcast "secure"+ usability, maintainablity, error control + watchdogs. hardware: terratec phase 22, amd 1400 mhz,512 RAM, 160 HDD, 2HE case connected to lan/dslrouter and balanced audio signal from mixer / pa. input:scheduler data via ical and/or dublin core metadata.analog balanced stereo audio in, 0 - 17 db outputpodcast xml, mp3 lame 128kbpsarchive quality luxury version: ogg, flacincluding appropriate id3 tags with event metadata.file repository (like apache-modmp3) with secure accesslogs and error reports control:via web interface proposal:when you do not have the time to master large amountsof recorded audio material by hand, an automatic gain controlat the time of the recording could help a lot.aiming at a good tradeoff between dynamic compression andsound quality, the main issue is to get a good leveled signalin the digital domain. it could be done by auto-readjusting the inputgain, or by using using the headroom of 8 bit,before the 24bit to 16 bit conversion for some smart compressor/limitermagic in real time. the mechanism of dithering/downsampling is known frommastering at the end of the chain (e.g. the waves l-1 maximizer).so why not using it for an unmaintained non-annoying AGC? maybe appropriate algorithms are available from voip projects,or by directly using vst plugins, or tuning of the jamin multibandcompressor. a window manager like gnome is not obligatory.we're experimenting with a plugin-chain foralsa, jackd, ecasound based on ladspa/fst.by now we can get a basic setup, but we're not evensure if the AGC is realizeable in this short time.. somebasic code is available but we'd need help withmaking it really run on a more advanced level.
