hi there, hope this mail is on topic on this list. for an FM and internet radio project during the worldcup 2006 in Berlin, we're looking for support on the linux audio developer side. it's a cultural project and we'll recording audio events at different places, from clubs to universitities. there's a small budget included and some preliminary code is available from an older streaming project. of course the project will be released under GNU. if you are interested to join in and help us with your work, 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 streaming of club events. your questions and comments are welcome. ..................................................................................................................... questions/tasks: - how to profit from 24bit to 16bit dithering and 88.2 khz to 44.1 kz resampling for 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/dsl router 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 output podcast xml, mp3 lame 128kbps archive quality luxury version: ogg, flac including appropriate id3 tags with event metadata. file repository (like apache-modmp3) with secure access logs and error reports control: via web interface proposal: when you do not have the time to master large amounts of recorded audio material by hand, an automatic gain control at the time of the recording could help a lot. aiming at a good tradeoff between dynamic compression and sound quality, the main issue is to get a good leveled signal in the digital domain. it could be done by auto-readjusting the input gain, or by using using the headroom of 8 bit, before the 24bit to 16 bit conversion for some smart compressor/limiter magic in real time. the mechanism of dithering/downsampling is known from mastering 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 multiband compressor. a window manager like gnome is not obligatory. we're experimenting with a plugin-chain for alsa, jackd, ecasound based on ladspa/fst. by now we can get a basic setup, but we're not even sure if the AGC is realizeable in this short time.. some basic code is available but we'd need help with making it really run on a more advanced level.